Field of the Disclosure
The technology of the disclosure relates generally to Web Real-Time Communications (WebRTC) interactive sessions.
Technical Background
Web Real-Time Communications (WebRTC) represents an ongoing effort to develop industry standards for integrating real-time communications functionality into web clients, such as web browsers, to enable direct interaction with other web clients. This real-time communications functionality is accessible by web developers via standard markup tags, such as those provided by version 5 of the Hypertext Markup Language (HTML5), and client-side scripting Application Programming Interfaces (APIs) such as JavaScript APIs. More information regarding WebRTC may be found in “WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web,” by Alan B. Johnston and Daniel C. Burnett (2012 Digital Codex LLC), which is incorporated herein in its entirety by reference.
WebRTC provides built-in capabilities for establishing real-time video, audio, and/or data streams in both point-to-point interactive sessions and multi-party interactive sessions. The WebRTC standards are currently under joint development by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF). Information on the current state of WebRTC standards can be found at, e.g., http://www.w3c.org and http://www/ietf.org.
To establish a WebRTC interactive session (e.g., a real-time video, audio, and/or data exchange), two web clients may retrieve WebRTC-enabled web applications, such as HTML5/JavaScript web applications, from a WebRTC application server. Through the web applications, the two web clients engage in a media negotiation to communicate and reach an agreement on parameters that define characteristics of the WebRTC interactive session. This media negotiation is known as a WebRTC “offer/answer” exchange. In an offer/answer exchange, a first web client operating on a sender computing device sends an “offer” to a second web client on a recipient computing device. The offer includes a WebRTC session description object that specifies media types and capabilities that the first web client supports and prefers for use in the WebRTC interactive session. The second web client then responds with a WebRTC session description object “answer” that indicates which of the offered media types and capabilities it supports and accepts for the WebRTC interactive session. Once the WebRTC offer/answer exchange is complete, the web clients may then establish a direct peer connection with one another, and may begin a real-time exchange of media or data. The peer connection between the web clients typically employs the Secure Real-time Transport Protocol (SRTP) to transport real-time media flows, and may utilize various other protocols for real-time data interchange.
However, a peer-to-peer architecture may not be optimal for real-time interaction in some circumstances. For instance, it may be necessary to provide recording or monitoring of the WebRTC interactive flow, and/or to address known incompatibilities between web clients at endpoints of the WebRTC interactive flow. The WebRTC standards provide for a WebRTC interactive flow to be routed through a TURN (Traversal Using Relay around NAT) server in situations in which a direct peer connection is not possible. However, a WebRTC interactive flow passing through a TURN server is encrypted, and thus cannot be examined or monitored at the TURN server. Moreover, converting a WebRTC interactive flow to a more easily manipulated format in real-time would require a gateway, which may be overly complex and/or commercially unavailable.